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Use AudioDeviceModule instead of TestAudioDeviceModule.

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This is step to allow migration of Test ADM to the AudioDeviceModuleImpl
as a base class to include AudioDeviceBuffer into SUT.

Also it will allow to remove WaitForRecordingEnd() method from Test
ADM

Bug: b/272350185, webrtc:15081
Change-Id: If2aa43ec0c31f6ad9aab8aa3e36cabc4a7a73c22
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300862
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39849}
This commit is contained in:
Artem Titov 2023-04-11 16:37:38 +02:00 committed by WebRTC LUCI CQ
parent 1b77daea81
commit fb8e3de0a8
10 changed files with 32 additions and 29 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

2
audio/BUILD.gn
View File

@ -134,6 +134,8 @@ if (rtc_include_tests) {
"../api/task_queue",
"../call:fake_network",
"../call:simulated_network",
"../modules/audio_device:audio_device_api",
"../modules/audio_device:audio_device_impl",
"../system_wrappers",
"../test:test_common",
"../test:test_support",

14
audio/test/audio_end_to_end_test.cc
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@ -16,16 +16,16 @@
#include "api/task_queue/task_queue_base.h"
#include "call/fake_network_pipe.h"
#include "call/simulated_network.h"
#include "modules/audio_device/include/test_audio_device.h"
#include "system_wrappers/include/sleep.h"
#include "test/gtest.h"
namespace webrtc {
namespace test {
namespace {
// Wait half a second between stopping sending and stopping receiving audio.
constexpr int kExtraRecordTimeMs = 500;
constexpr int kSampleRate = 48000;
} // namespace
AudioEndToEndTest::AudioEndToEndTest()
@ -54,8 +54,8 @@ AudioEndToEndTest::CreateRenderer() {
}
void AudioEndToEndTest::OnFakeAudioDevicesCreated(
TestAudioDeviceModule* send_audio_device,
TestAudioDeviceModule* recv_audio_device) {
AudioDeviceModule* send_audio_device,
AudioDeviceModule* recv_audio_device) {
send_audio_device_ = send_audio_device;
}
@ -81,11 +81,5 @@ void AudioEndToEndTest::OnAudioStreamsCreated(
receive_stream_ = receive_streams[0];
}
void AudioEndToEndTest::PerformTest() {
// Wait until the input audio file is done...
send_audio_device_->WaitForRecordingEnd();
// and some extra time to account for network delay.
SleepMs(GetSendTransportConfig().queue_delay_ms + kExtraRecordTimeMs);
}
} // namespace test
} // namespace webrtc

13
audio/test/audio_end_to_end_test.h
View File

@ -16,6 +16,8 @@
#include "api/task_queue/task_queue_base.h"
#include "api/test/simulated_network.h"
#include "modules/audio_device/include/audio_device.h"
#include "modules/audio_device/include/test_audio_device.h"
#include "test/call_test.h"
namespace webrtc {
@ -26,7 +28,7 @@ class AudioEndToEndTest : public test::EndToEndTest {
AudioEndToEndTest();
protected:
TestAudioDeviceModule* send_audio_device() { return send_audio_device_; }
AudioDeviceModule* send_audio_device() { return send_audio_device_; }
const AudioSendStream* send_stream() const { return send_stream_; }
const AudioReceiveStreamInterface* receive_stream() const {
return receive_stream_;
@ -39,9 +41,8 @@ class AudioEndToEndTest : public test::EndToEndTest {
std::unique_ptr<TestAudioDeviceModule::Capturer> CreateCapturer() override;
std::unique_ptr<TestAudioDeviceModule::Renderer> CreateRenderer() override;
void OnFakeAudioDevicesCreated(
TestAudioDeviceModule* send_audio_device,
TestAudioDeviceModule* recv_audio_device) override;
void OnFakeAudioDevicesCreated(AudioDeviceModule* send_audio_device,
AudioDeviceModule* recv_audio_device) override;
void ModifyAudioConfigs(AudioSendStream::Config* send_config,
std::vector<AudioReceiveStreamInterface::Config>*
@ -50,10 +51,8 @@ class AudioEndToEndTest : public test::EndToEndTest {
const std::vector<AudioReceiveStreamInterface*>&
receive_streams) override;
void PerformTest() override;
private:
TestAudioDeviceModule* send_audio_device_ = nullptr;
AudioDeviceModule* send_audio_device_ = nullptr;
AudioSendStream* send_stream_ = nullptr;
AudioReceiveStreamInterface* receive_stream_ = nullptr;
};

6
audio/test/audio_stats_test.cc
View File

@ -17,6 +17,9 @@ namespace webrtc {
namespace test {
namespace {
// Wait half a second between stopping sending and stopping receiving audio.
constexpr int kExtraRecordTimeMs = 500;
bool IsNear(int reference, int v) {
// Margin is 10%.
const int error = reference / 10 + 1;
@ -41,7 +44,8 @@ class NoLossTest : public AudioEndToEndTest {
void PerformTest() override {
SleepMs(kTestDurationMs);
send_audio_device()->StopRecording();
AudioEndToEndTest::PerformTest();
// and some extra time to account for network delay.
SleepMs(GetSendTransportConfig().queue_delay_ms + kExtraRecordTimeMs);
}
void OnStreamsStopped() override {

3
audio/test/low_bandwidth_audio_test.cc
View File

@ -57,7 +57,8 @@ class AudioQualityTest : public AudioEndToEndTest {
// Let the recording run for a small amount of time to check if it works.
SleepMs(1000);
} else {
AudioEndToEndTest::PerformTest();
// Sleep for whole audio duration which is 5.4 seconds.
SleepMs(5400);
}
}

2
modules/audio_device/include/test_audio_device.cc
View File

@ -439,7 +439,7 @@ size_t TestAudioDeviceModule::SamplesPerFrame(int sampling_frequency_in_hz) {
return rtc::CheckedDivExact(sampling_frequency_in_hz, kFramesPerSecond);
}
rtc::scoped_refptr<TestAudioDeviceModule> TestAudioDeviceModule::Create(
rtc::scoped_refptr<AudioDeviceModule> TestAudioDeviceModule::Create(
TaskQueueFactory* task_queue_factory,
std::unique_ptr<TestAudioDeviceModule::Capturer> capturer,
std::unique_ptr<TestAudioDeviceModule::Renderer> renderer,

2
modules/audio_device/include/test_audio_device.h
View File

@ -82,7 +82,7 @@ class TestAudioDeviceModule : public AudioDeviceModule {
// `renderer` is an object that receives audio data that would have been
// played out. Can be nullptr if this device is never used for playing.
// Use one of the Create... functions to get these instances.
static rtc::scoped_refptr<TestAudioDeviceModule> Create(
static rtc::scoped_refptr<AudioDeviceModule> Create(
TaskQueueFactory* task_queue_factory,
std::unique_ptr<Capturer> capturer,
std::unique_ptr<Renderer> renderer,

1
test/BUILD.gn
View File

@ -1099,6 +1099,7 @@ rtc_library("test_common") {
"../call:simulated_network",
"../call:simulated_packet_receiver",
"../call:video_stream_api",
"../modules/audio_device:audio_device_api",
"../modules/audio_device:audio_device_impl",
"../modules/audio_mixer:audio_mixer_impl",
"../modules/rtp_rtcp",

8
test/call_test.cc
View File

@ -22,6 +22,8 @@
#include "call/fake_network_pipe.h"
#include "call/packet_receiver.h"
#include "call/simulated_network.h"
#include "modules/audio_device/include/audio_device.h"
#include "modules/audio_device/include/test_audio_device.h"
#include "modules/audio_mixer/audio_mixer_impl.h"
#include "rtc_base/checks.h"
#include "rtc_base/event.h"
@ -770,9 +772,9 @@ std::unique_ptr<TestAudioDeviceModule::Renderer> BaseTest::CreateRenderer() {
return TestAudioDeviceModule::CreateDiscardRenderer(48000);
}
void BaseTest::OnFakeAudioDevicesCreated(
TestAudioDeviceModule* send_audio_device,
TestAudioDeviceModule* recv_audio_device) {}
void BaseTest::OnFakeAudioDevicesCreated(AudioDeviceModule* send_audio_device,
AudioDeviceModule* recv_audio_device) {
}
void BaseTest::ModifySenderBitrateConfig(BitrateConstraints* bitrate_config) {}

10
test/call_test.h
View File

@ -26,6 +26,7 @@
#include "api/units/time_delta.h"
#include "api/video/video_bitrate_allocator_factory.h"
#include "call/call.h"
#include "modules/audio_device/include/audio_device.h"
#include "modules/audio_device/include/test_audio_device.h"
#include "test/encoder_settings.h"
#include "test/fake_decoder.h"
@ -271,8 +272,8 @@ class CallTest : public ::testing::Test, public RtpPacketSinkInterface {
std::vector<RtpExtension> rtp_extensions_;
rtc::scoped_refptr<AudioProcessing> apm_send_;
rtc::scoped_refptr<AudioProcessing> apm_recv_;
rtc::scoped_refptr<TestAudioDeviceModule> fake_send_audio_device_;
rtc::scoped_refptr<TestAudioDeviceModule> fake_recv_audio_device_;
rtc::scoped_refptr<AudioDeviceModule> fake_send_audio_device_;
rtc::scoped_refptr<AudioDeviceModule> fake_recv_audio_device_;
};
class BaseTest : public RtpRtcpObserver {
@ -290,9 +291,8 @@ class BaseTest : public RtpRtcpObserver {
virtual std::unique_ptr<TestAudioDeviceModule::Capturer> CreateCapturer();
virtual std::unique_ptr<TestAudioDeviceModule::Renderer> CreateRenderer();
virtual void OnFakeAudioDevicesCreated(
TestAudioDeviceModule* send_audio_device,
TestAudioDeviceModule* recv_audio_device);
virtual void OnFakeAudioDevicesCreated(AudioDeviceModule* send_audio_device,
AudioDeviceModule* recv_audio_device);
virtual void ModifySenderBitrateConfig(BitrateConstraints* bitrate_config);
virtual void ModifyReceiverBitrateConfig(BitrateConstraints* bitrate_config);